There are a variety of known protocols for establishing media stream communications, such as voice, data, video, or combinations thereof, over an IP network. Protocols for establishing media stream communications over an IP network are referred to herein as IP telephony protocols. One example of an IP telephony protocol is the media gateway control protocol (MGCP). MGCP defines signals and events by which a software entity, known as a media gateway (MG), is controlled by another software entity, known as a media gateway controller (MGC), in a packet network. The media gateway controller processes call control signaling from one or more signaling gateways (SGs) and utilizes MGCP media control signaling to establish media streams between MGs. An MGC that processes call control signaling in this manner is also referred to as a call agent. The terms media gateway controller and call agent are used interchangeably herein. The media gateway controller performs call control functions, such as translations, resource management, media capabilities negotiation and selection, and media stream management. It can also provide additional services.
FIG. 1 illustrates conventional communications using MGCP. In FIG. 1, MGC 100 receives call signaling information from SGs 102 and 103 and controls MGs 104 and 105 to establish packetized media stream communications between end users in packet network 106. For example, SG 102 and MG 104 can be associated with a calling party end user device for a given media stream communication. Similarly, SG 103 and MG 105 can be associated with a called party end user device for a given media stream communication. MGC 100 can control MGs 104 and 105 to establish media stream communications between the called and calling end user devices, such as PSTN terminals.
A detailed explanation of MGCP is found in Media Gateway Control Protocol (MGCP), Version 0.1 Draft, Internet Engineering Task Force, Feb. 21, 1999, the disclosure of which is incorporated herein by reference in its entirety.
Another example of an IP telephony protocol is International Telecommunications Union (ITU) Recommendation H.323. H.323 defines a protocol by which endpoints, such as gateways, terminals, or multipoint control units MCUs), can place calls in a packet network. A gateway translates between circuit-switched and packet-switched communication protocols. A terminal is a device, such as an IP terminal, that provides end user access to a network. An MCU is a device that supports conferences between three or more endpoints. H.323 defines a gatekeeper as an entity that provides address translation and controls access to the packet network for H.323 endpoints. The gatekeeper can also provide additional services, such as call control and supplementary services.
FIG. 2 illustrates an example of conventional H.323 communications. In FIG. 2, a first gateway 200 can be associated with a calling end user device and a second gateway 202 can be associated with a called end user device for a given media communication. Gatekeeper 204 performs call signaling functions, such as call setup and teardown, to establish calls between end user devices associated with gateways 200 and 202. The end user devices can be PTSN terminals connected to gateway 200. Alternatively, gateway 200 can be omitted and replaced by H.323 terminals or H.323 MCUs. Once gatekeeper 204 performs the call signaling functions necessary to set up a call, the media stream for the call flows between gateways 200 and 202. Detailed information relating to H.323 can be found in ITU Recommendation H.323, Packet Based Multimedia Communications Systems, February 1998, the disclosure of which is incorporated herein by reference in its entirety.
Yet another IP telephony protocol is ITU Recommendation H.248. The Internet Engineering Task Force (IETF) formed the MEGACO Group to evolve the MGCP protocol. As the MEGACO Group matured the protocol, the MEGACO Group allied itself with the ITU, and the specification developed by the MEGACO Group has become known as ITU Recommendation H.248. Thus, ITU recommendation H.248 can be viewed similarly to MGCP.
Another IP telephony protocol is the session initiation protocol (SIP). SIP is an application layer signaling protocol for creating, modifying, and terminating sessions between one or more participants. The sessions include internet multimedia conferences, internet telephone calls, and multimedia distribution. SIP originated from Columbia University and is gaining acceptance as a protocol for exchanging call signaling information over a packet network. A detailed description of SIP can be found in Request for Comments (RFC) 2543 SIP: Session Initiation Protocol, March 1999, the disclosure of which is incorporated herein by reference in its entirety.
In addition to the published protocols described above, many vendors of telecommunications equipment and services are supporting IP telephony applications via proprietary protocols.
All of the IP telephony protocols described above are being implemented by various vendors. However, standards for interworking equipment that communicates using one protocol with equipment that communicates using another protocol are immature, nonexistent, or focus only on a specific type of application. Accordingly, there exists a long-felt need for a novel method and apparatus for interworking between IP telephony protocols.